Telecommunication services for voice and multimedia (e.g., audio, video, and data) have traditionally been provided using networking technologies such as public switched telephone networks (PSTN). Typically, in such networks, voice signals are converted to digital form and time division multiplexing (TDM) is used to allocate different conversations to periodic time slots. More recently, other networking technologies have been used to carry voice and multimedia information. Such technologies include internet protocol (IP), a formal set of rules for sending data across the internet; frame relay (FR), a telecommunication service providing variable-sized data packets for cost-efficient data transfer; and asynchronous transfer mode (ATM), a high speed networking standard. Such networks provide a single, common and shared infrastructure, thus flexibly enabling a potentially wide variety of new applications and services for customers.
Networks using these technologies employ a variety of call control services using a variety of protocols, for example, integrated services digital network user part (ISUP)-over-TDM, H.323-over-IP and session initiation protocol (SIP)-over-IP. The H.323 standard is a comprehensive and very complex suite of protocols that provide specifications for real-time, interactive videoconferencing, data sharing and audio applications such as IP telephony. Alternatively, the SIP protocol is advantageous as it is a streamlined protocol developed specifically for IP telephony in view of popular web-based services. More efficient than H.323, SIP takes advantage of existing protocols to handle certain parts of the call control process. For example, Media Gateway Control Protocol (MGCP), or H.248 protocol, is used between the signaling call control entity and the media gateway entity in a master-slave scheme, while SIP works as a peer-to-peer protocol between the signaling entities (e.g., call entity of MGCP or H.248) along with an indication of what media needs to be used by the media entities. Therefore, SIP can take the advantage of the master-slave protocol like MGCP or H.248 that is being used to provide the media control function of the devices satisfying the needs of SIP in order to set up the session. It may be noted that the media devices may be connected to the circuit-switched based networks, like PSTN.
Regardless the standard, as traditional PSTN networks migrate toward other networking technologies and protocols they must interface with networks using differing protocols (e.g., traditional PSTN networks). However, developing a single common cost-efficient architecture for real-time communication services for audio, video, and data that supports multiple existing call control protocols (e.g., PSTN, SIP and H.323) is difficult.
Many telecommunications providers are selecting IP as the access technology for new telecommunications networks that carry voice, data and multimedia information. Such networks are often referred to as Voice over IP (VoIP) networks. Many such providers are using a core network that uses the SIP protocol for signaling and call flow operations. Such SIP-based VoIP networks are advantageous in that they use the same access technology (IP) as many other networks, such as the Internet, which facilitates transmitting information to a wide range of destinations. However, while SIP-based telecommunications networks are advantageous, not all networks and components support SIP. Different call control mechanisms/signaling protocols over the IP networks may use different communication protocols. Therefore, there is a need to permit SIP networks to coexist with more traditional networks, such as circuit-switched networks, and/or IP networks operating with a different protocol.
To address this need, some IP telecommunications networks rely on nodes referred to herein as border elements (BEs) to provide an interface between a a customer's premises into the VoIP network infrastructure. Such BE's are typically used to translate between the protocol of a customer network and the SIP protocol used by the VoIP network as a common call control signaling protocol among different functional entities of the VoIP common infrastructure. These BEs also perform a wide variety of other functions, including signaling, call admission control, media control and security functions.